The purpose and function of your VoIP Gateway is to provide an interface involving the traditional telephone networks using digital TDM (Time Division Multiplexing) technology and that of IP Networks made to carry IP packets containing digital speech. The VoIP Gateway has to translate the digital media format applied to the neighborhood network and the PSTN (Public Switched Telephone Network) both in directions. Furthermore, the Gateway may also have to translate between your different signalling protocols applied to the area network and PSTN to VoIP technology.Today’s Voice over IP networks digitise analogue speech for calls my method of a Codec which stands for Coder Decoder or Compressor De-compressor. There are a number of codecs used within VoIP systems, most abundant in common of these being ITU-T codecs G.711 and G.729, and also the GSM codecs used inside the mobile telephone networks. Two versions from the g.711 codec are employed inside traditional digital telephone networks which are based largely on 64Kbps digital telephone channels. Digital telephone channels are multiplexed together utilizing a method called Time Division Multiplexing to make either T1 or E1 lines. A T1 lines are made to carry 24 digital phone calls, whereas an E1 line is made to carry 30 digital calls. These multiplexed trunk line is accustomed to interconnect exchanges and also connect a traditional PBX (Private Branch Exchange) to a Telephone Operator’s exchange.A VoIP Gateway will need to support a number of codecs, so as to be capable of translate between the digital codec formats utilized by the PSTN as well as the VoIP enabled Lan.In addition to translating digital voice formats, the gateway must also be able to translate between different signalling formats. The Public Switched Telephone Network uses two main types of signalling which can be called CAS (Channel Associated Signalling) and CCS (Common Channel Signalling). The previous is used on the analogue phone line between the telephone user and the exchange by passing telephone dialling codes often by way of audio tones, which method is known as DTMF (Dual Tone Multi Frequency). Pulse dialling is also found in some systems. Common Channel Signalling is used between exchanges and gateways to give telephone signalling information inside wider telephone network.SIP (Session Information Protocol) has evolved since the dominant signalling protocol within VoIP Systems, and has largely replaced the cumbersome ITU signalling protocol H.323. SIP is an easy client / server protocol that is used to create, maintain and teardown VoIP telephone calls. MGCP (Media Gateway Control Protocol) is used by some Voip systems to manage what of a VoIP Gateway and Cisco uses SCCP (Skinny Call Control Protocol) on some older systems as a method of communicating between your customers as well as the Cisco Callmanager call control agent.Signalling System 7 is the CCS system used within all of the Public Telephone Networks, including land-based and mobile systems. In reality, frequently it’s called the “Glue” that links all Public networks together and allows us to make calls between Cellphones, land-based phones and VoIP phones.